Anyone interested in a digital amplifier project?

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Brian Brown said:
"But the final 11dB represents digital gain and can, depending on the peak level represented by the incoming digital data, cause the amp to crash into high distortion. Believe it or not, TACT was obliged to introduce this extra gain because various dealers were unsettled by the fact that the volume control could be advanced to full without creating distorted sound. "
...
Brian.:cubist: [/B]

:D Now the marketing droids can talk about an amp that has a volume control going all the way to 11! ;)
 
Why mains->smooth->switchmode->smooth->class-d->smooth->speaker?

Why not just mains->smooth->class-d->smooth->speaker.

Yeh im thinking aloud, and yeh i know it probably cant be done... but wouldnt it be nice?

Anyone want to build up some high impedance drivers to do this? ;)
 
Re: Equibit Experiment

Brian Brown said:
Most Class D amps require feedback for stability

most class D amps become highly unstable with the application of negative feedback. this is the main problem we face with class D in general. they are now trying to use DSP (hence the phrase digital amp / class T, ect...) in an effort to linearize the output without feedback. whether or not the "digital" amplifier accepts a digital signal or analog signlal at the input, the DSP employed is there to provide a "amplifier with feedback" like qualities without the application of real feedback.

there are currently some very interesting patents on class BD amplifier technology (which has nothing to do with traditional class B amplifiers). it employs a truely ingenious method of triple rail switching (-V / gnd / +V) at the output with only a single inductor. this allows application of some traditional negative feedback since the amplifier output is truely resting at a physical ground and there is a huge reduction in switching noise. damping factor is also increased significantly. all of this without DSP =). i believe high power class BD (and similar triple output switching systems) will eventually be the solution to the poor s/n, distortion performance, DF, and frequency responce of current class D and AD technology. combined with some DSP the possibilities are endless.
 
Re: This is all fine and dandy but ....

Petter said:
To get optimal performance we need a variable power supply. TacT does this with SMPS. How do you propose we do it?

Petter

This is another area that I'll be playing with a bunch of options once my first board is functioning.

A couple of points:

TacT uses a variable supply as a form of volume control. Once the volume is set, the supply voltage remains constant. The supply voltage isn't varying dynamically with the music signal like a Class H amp does. With the volume set, the available dynamic range is primarily determined by the digital word size. (The original Millennium used 18bits, the current version uses 24.)

The TAS5012 has a dynamic range of 102db with a fixed supply. The TAS5015 has a dynamic range of 112db with a fixed supply. There's probably very few linear amps that can approach these numbers in the real world. My thinking is that as long as the supply voltage of an Equibit amp correctly matches the amp's peak power to the speaker's capabilities, there probably isn't much to be gained :xeye: from further using the supply voltage as a volume control. As the previously quoted article showed, TacT was willing to throw away 11db of their digital resolution to marketing. I have a hard time believing they would do that if they thought it would have an impact on sound quality.

Another point is that TacT doesn't make their amps for one specific speaker. They need a user variable power supply so that the amp can be matched to speakers with different efficiencies and power handling capabilities.

My intention is to match the power supply voltage to the speaker the amp is driving. So my power supply options will require the ability to set the voltage to a specific value, but they won't necessarily have to be continuously variable.

*******

My first digital amp board doesn't have the front-end supply included. It does have the main bus caps (separate for each channel) and a relay to aid in power up and power down sequencing. There are footprints for LT1083 voltage regulators for each channel's supply that I can either use or bypass. Per TI's guidelines, there are snubber circuit footprints at each node in the power input and output chains. The values for these snubbers will be empirically tweaked after the board is built and operating.

Ultimately, I'm hoping to come up with a good off-line switching power supply (possibly with power-factor correction) with feedback to maintain 1.0V to 1.5V across linear post-regulators. This architecture should have the best combination of efficiency, compactness, and dynamic performance. If desired, it would also easily be adapted to continuously variable voltage using digital control.

The linear post-regulators should help make the supply very stiff in the audio band. This should allow me to use some fairly heavy LC filtering at the switcher output to get rid of any high frequency trash. Without the linear post-regulators, heavy LC filters would limit the switcher's response bandwidth to the point that it would get down into the audio range. (A side benefit of the LC filters is that they help limit the precharge inrush current of the bus caps.)

As previously discussed, Equibit amps need a very stiff power supply with low noise in the audio band because it's open loop. The power supply is analogous to a voltage reference in a DAC. There is zero power supply rejection ratio in the audio band. Ultrasonic power supply noise is a completely different matter. By its very nature, an Equibit amp produces 384KHz switching trash. Most switchers operate in the 100KHz to 500KHz range (I'd probably want to use at least 250KHz). The literature isn't very clear if there's a concern with the 384KHz intermodulating with 250KHz to produce products down into the audio band. If this isn't an issue, then the heavy LC filters and linear post-regulators could be eliminated. This would allow the response bandwidth of the switcher to be increased enough to make the audio band very stiff.

Another approach would be to make the switcher run synchronous to the amp's 384KHz carrier. This would ensure that there would be no intermodulation products.

*******

There's no reason that an Equibit amp won't work with a more traditional supply consisting of a line transformer / rectifier / capacitor arrangement. It will work just as well as it would for a linear amp. Of course it will be larger and more inefficient, but I suspect that isn't much of a concern to most of the DIY community.

Just as with a linear amp, the challenge is to make this supply as stiff as possible with a super low impedance at audio frequencies.
In fact, the super high damping factor of an Equibit amp will reveal the quality of the power supply even more. The Equibit output stage directly shunts any back EMF from the speaker straight into the supply. This doesn't mean that an Equibit amp will have higher distortion with a given supply. Rather it means that the elimination of distortion components resulting from the back EMF influencing a linear amp's output drivers will make it easier to hear what's happening in the supply.

High frequency power supply trash is a big concern with linear amps. I believe it will be less so with an Equibit amp.

A basic transformer / rectifier / capacitor supply is easy to design, and there's low risk of problems. This makes it a safer choice for many designs.

A super low impedance, tightly regulated conventional supply is a completely different matter. IMO these are very difficult to do well. I think that the quality of a conventional high-end linear amp is more likely to hinge on the quality of its power supply than on the quality of its output stage.

A high quality switching power supply is a tricky proposition, no doubt. But I think that for an Equibit amp it will be easier to achieve the best results from a switching supply than from a conventional one.

Once again, I'm planning to try both.

Just to be clear, I don't consider myself a switching power supply guru, especially when it comes to magnetics. But I have been involved with them for a number of years and have spent quite a bit of time troubleshooting system noise and performance issues.

More importantly, I've got some friends who are transcendent gurus with these things that I can turn to if I run into issues.

******

Another approach that I'm looking at is to use a stack of gell-cell batteries in series (one for each channel). With bus caps in parallel, this should be a very low impedance stiff supply. I'll probably try them with and without linear regulators. (Using batteries direct, the voltage can be controlled by the number of cells in series.)

This approach wouldn't fly for most commercial applications, but DIY is a completely different thing.

The high efficiency of Equibit amps makes batteries much more practical to consider for a power amplifier.

******

To initially get my amp up and running, I'm planning on using a commercial switching power brick.

Regards,
Brian.:cubist:
 
Would be a variable power supply with OPA549 (page 13 of OPA549 datasheet) usable? How would it react to a capacitive load? Would it cause the OP-AMP oscillate?

You mentioned that an Equibit amp has super high damping factor, but in the TAS5100EVM Data Report (sleu011a document) it is stated that the damping factor is only 15 (1kHz, 8 ohm). That is far away from super.

Regards,
Dejan
 
Damping Factor

You mentioned that an Equibit amp has super high damping factor, but in the TAS5100EVM Data Report (sleu011a document) it is stated that the damping factor is only 15 (1kHz, 8 ohm). That is far away from super.

I probably shouldn't have used the specific phrase 'damping factor', but the concept being considered is the same.

Normally a damping factor of 15 would be a downright lousy figure. But this is an issue of the term's definition and the way it is traditionally measured.

Damping factor is a figure of merit to indicate how well an amplifier can control a load. In terms of how well an Equibit amp can do this in a real-life situation driving an actual speaker with lots of electrically and mechanically induced back-EMF, it is superior to most traditional linear amp designs.

(Note that the following discussion is presuming a very stiff, wide-bandwidth, high-quality power supply).

A linear amp with bipolar output devices is usually highly susceptible to back-EMF, which will alter the conduction of the output devices.

A linear amp with MOSFET output devices is much better in this regard, because the MOSFETs don't react to the back-EMF, and it doesn't affect the actual output of the amp the way it does with bipolar devices. However, the MOSFETs are only partially turned on, so there is considerable impedance between the back-EMF and the power supply.

A switching amplifier, because its output drivers are either fully turned on or off, will directly shunt the back-EMF into the power supply. There is only the Rds-on of the MOSFETs and the impedance of the low-pass output reconstruction filter to overcome (this is one reason that the output inductors should have as low of series resistance as possible, preferable below 50mOhms.)

Strictly speaking, damping factor is defined as the ratio of the load impedance to the amp's output impedance. It is given as a single number with a given test frequency at a certain load resistance. It is also measured with a dummy test load resistor!

Let's talk about negative feedback. Most of you are probably familiar with the way that negative feedback can be used to 'force' the output to have very low THD measurements when driving a resistive load, but then causes all sorts of stability issues and distortion with an actual speaker. This is a similar issue with damping factor measurements.

Negative feedback lowers the effective output impedance of an amplifier, especially into a load resistor. Back-EMF, by its very nature, contains lots of high frequency trash that most negative feedback loops don't have the response bandwidth to keep under control. This can mean that the low effective output impedance into a static load is lost in a real life situation. Even worse, the back-EMF can get into the feedback loop and further screw up the amplifier's output. This is the exact opposite of what a high damping factor is supposed to represent!

This is why an Equibit amplifier, running open-loop with zero negative feedback, scores poorly when testing damping factor with a dummy load.

In real life, an Equibit amplifier has no feedback at all to be screwed up by the back-EMF, and it has a very direct low-impedance path between the power supply and speaker that is more effective at shunting out back-EMF.

So in terms of the 'Spirit of the meaning' of damping factor, an Equibit amplifier is superior to most others.

I'm surprised that TI included this spec in the report without any explanation.

Regards,
Brian.:cubist:
 
Re: OPA549

Rookie said:
Would be a variable power supply with OPA549 (page 13 of OPA549 datasheet) usable? How would it react to a capacitive load? Would it cause the OP-AMP oscillate?
This is an interesting idea. I especially like that the OPA549 can operate up to 60V.

I wouldn't want to use any type of linear regulator as the sole means of voltage adjustment. The voltage drop would be considerable, and it'd take a pretty hefty heat sink to dissipate the waste heat. That's why I was proposing a switching preregulator to keep the voltage drop across the linear regulator to a minimum.

The stability into a capacitive load is certainly a concern. I think this could be worked out, but it'd be a little tricky.

By comparison, an LT1083 is more efficient, probably lower noise in this application, and is stable into unlimited capacitive loads. It does have a 30V input limitation, however.
 
Re: Class BD

R. McAnally said:


there are currently some very interesting patents on class BD amplifier technology (which has nothing to do with traditional class B amplifiers). it employs a truely ingenious method of triple rail switching (-V / gnd / +V) at the output with only a single inductor. this allows application of some traditional negative feedback since the amplifier output is truely resting at a physical ground and there is a huge reduction in switching noise. damping factor is also increased significantly. all of this without DSP =). i believe high power class BD (and similar triple output switching systems) will eventually be the solution to the poor s/n, distortion performance, DF, and frequency responce of current class D and AD technology. combined with some DSP the possibilities are endless.

We're only beginning to see what I'm sure will be a flood of new switching amplifier techniques. While most efforts seem to be focusing on low cost, small size, and high efficiency, there's lots of stuff coming to raise the bar on sound quality. I'm very excited about the prospects.

I just hope that we, as individuals, get access to this new stuff.

It seems like so many of the new technologies coming out are arriving with licensing agreements that only large companies can afford. This isn't an issue only to hobbiests, but also to small audio manufacturers.

Mitsubishi is now producing a digital amp chip (M65817AFP) that works with either IIS PCM data or DSD data from an SACD. Apparently they're making it for Sony. I sure wish I could get my hands on some of these to play with!

I can't say enough about how happy I am that I can get the TI chips!

Brian.:cubist:
 
Helo,


I am new to this forum. I have just started drawing shematics for TAS5015 based amplifier. If the design will be any good I am willing to share it.

My plan is as follows:

Run TAS5015 from external oscillator, maybe even 100MHz (though I have found swiss firm that produces 98.304Mhz oscillators http://www.microcrystal.com/Products/TechDoku/MCSOHVT.pdf ) and use AD1895 or AD1896 asynchronous sample rate converter in front. All incoming data would be upsampled to 24 bit 192 khz and TAS5015 run in master mode.

Build discrete output stage with supply voltage up to 100V for the first design and maybe higher in next iteration. Output FETs would be Fairchild FDD3672 (44A, 28mE) which have very low gate charge, comparable to IRF520. Gate driver will be either HIP2100 from Intersil or discrete design using miniature pulse transformers (2.5mm ferrite toroids) with differential transformer between upper and lower gate driver to maintain proper dead time. I think this design would work pretty well, since I already tried it in analog class D amplifier.

I think it is better to optimize dead time (non overlap of conduction times of upper and lower FET in H-bridge) on driver and not to rely on PWM modulator. Anyhow, I think that there is actually no dead time between PWM_AP and PWM_AN outputs of any TAS50xx chip. Look at page 7 Figure1 of TAS5182 datasheet.

I also think it would be a good idea to physically separate modulator and output bridge section. I intend to use some Cat5 UTP cable and use LVDS chips to transmit and receive PWM information. If TAS5015 outputs are trully differential it would be even possible to use some resistive dividers on transmitting size.

I think separate approach has several merits: when new modulator comes along, it is necessary only to change modulator section, power section remains the same. I think also new developement will be in use of multiphase designs (look at Philips patent application "20020053945 Switching power amplifier"). It is relatively easy to add new phases with modular design. It may be even possible to use two TAS5015 modulators runing at 96kHz with skewed LRCLK. (just an idea, haven't look at it yet).

Advantage of multiphase design is: lower current in each halfbridge switching stage, lower switching frequency, ripple cancelation on output LC filter. I think also TI has found this since i suppose there is already some skew between left and right channel and also between A and B half of H-Bridge. (US patent 6373336 and US patent application "20020060605 Amplifiers").

Regarding power supply: I will use switching power supply with variable output voltage. Like in TACT design advantage is increased dynamic range and there is also less switching noise at low voltage and low volume.

Regarding dynamics of the power supply: switching power supplies can be made very fast. Look at the current multiphase designs for modern processors. So the best way would be to have isolated PFC conveter with large bank of capacitors on the secondary and relatively slow regulation loop (as PFC is supposed to do) and then multiphase synchronous buck converter. This would look almost exactly as already existing H-bridges and use same transistors and driver circuits.

Other possibility would be to use relatively simple power supply (mains transformer and rectifier with large capacitor bank) and use PEDEC (pulse edge delay error correction) by Niels Karsten.http://bogo-united.oeb.tdk.net/graphics/powerhouse/user_graphics/library/phd_volume_1/chapter_9.pdf. There is also similar technology to PEDEC http://pearlx.snu.ac.kr/Publication/PESC0203.pdf and maybe I can also think of something similar.

As I can see this project will take most of my free time next year.
Before I will make PCbs I would like to have some positive confirmation about timing of PWM outputs of TAS5015 chip. Is there actually no dead time between upper and lower PWM output and if there is actually time skew between left anf right channel and between left and right half of the each channel"s H-bridge? What would be best SPDIF receiver chip for the modulator and is there any advantage in using DF1704 oversampling filter?
Can anyone answer ?

Best regards,

Jaka Racman
 
Digital filter is unlikely to be of much use as there is already some significant math going down in the TAS5015.

There is a companion chip which has been previewing on the TI site for a very long time which takes what I understand to be the balanced outputs (which have no skew) and sets up the rest of the bridge. I guess what I am trying to say is that there are two balanced signals controlling the entire bridge, and I believe this is what you asked to know. If you look at the lesser chips, they have single ended outputs performing the same function.

Petter
 
Re: Re: Class BD

Brian Brown said:
We're only beginning to see what I'm sure will be a flood of new switching amplifier techniques. While most efforts seem to be focusing on low cost, small size, and high efficiency, there's lots of stuff coming to raise the bar on sound quality. I'm very excited about the prospects.

I just hope that we, as individuals, get access to this new stuff.

Currently Hafler / Rockford holds the patent on the most recent class BD incarnation and i believe they sell them mostly in the pro-audio scene. The best part is there is no "all in one chip" for this type of technology. no DSP is required. they brought feedback back into the class D architecture and that took care of most everything as well or even better than DSP, without sacraficing stability.
 
Jaka Racman said:
I am new to this forum. I have just started drawing shematics for TAS5015 based amplifier. If the design will be any good I am willing to share it.

Hi!

I'm glad to hear you're working on this!

My plan is as follows:

Run TAS5015 from external oscillator, maybe even 100MHz (though I have found swiss firm that produces 98.304Mhz oscillators http://www.microcrystal.com/Products/TechDoku/MCSOHVT.pdf ) and use AD1895 or AD1896 asynchronous sample rate converter in front. All incoming data would be upsampled to 24 bit 192 khz and TAS5015 run in master mode.

Build discrete output stage with supply voltage up to 100V for the first design and maybe higher in next iteration. Output FETs would be Fairchild FDD3672 (44A, 28mE) which have very low gate charge, comparable to IRF520. Gate driver will be either HIP2100 from Intersil or discrete design using miniature pulse transformers (2.5mm ferrite toroids) with differential transformer between upper and lower gate driver to maintain proper dead time. I think this design would work pretty well, since I already tried it in analog class D amplifier.

BEAUTIFUL.

I think it is better to optimize dead time (non overlap of conduction times of upper and lower FET in H-bridge) on driver and not to rely on PWM modulator. Anyhow, I think that there is actually no dead time between PWM_AP and PWM_AN outputs of any TAS50xx chip. Look at page 7 Figure1 of TAS5182 datasheet.

The PCM to PWM modulator chips (such as the TAS5012 and TAS5015) don't introduce any dead time to their PWM outputs.

The TAS5110 integrated H-bridge and the TAS5182 H-bridge driver chips DO introduce dead time. They do this by monitoring the gate charge of the output MOSFETs. This allows them to compensate for varying temperature and load characteristics. The dead time is adjusted by means of external resistors (Rdt - one for the high side of the bridge and one for the low side). In the case of the TAS5182, these are connected to pins 6 and 7, respectively.

From your initial description I'm presuming that you're planning to connect your output circuit directly to the TAS5015. In this situation you will need to implement your own dead-time control. (This probably is a better way to do it. I'm not going to be quite as ambitious in my first attempt, so I'm going to rely on TI's output chips.)

I also think it would be a good idea to physically separate modulator and output bridge section. I intend to use some Cat5 UTP cable and use LVDS chips to transmit and receive PWM information. If TAS5015 outputs are trully differential it would be even possible to use some resistive dividers on transmitting size.

I can agree with you on the benefits of separating the output section, although this is another thing I won't be doing on my first attempt.

The TAS5015 outputs ARE NOT differential. They're single ended LVTTL referenced to ground. They have four separate lines for each transistor in the H-bridge because the timing between the uppers and lowers of the pairs is skewed.

I think separate approach has several merits: when new modulator comes along, it is necessary only to change modulator section, power section remains the same. I think also new development will be in use of multiphase designs (look at Philips patent application "20020053945 Switching power amplifier"). It is relatively easy to add new phases with modular design. It may be even possible to use two TAS5015 modulators runing at 96kHz with skewed LRCLK. (just an idea, haven't look at it yet).

I'm not sure that the interface for a multiphase design would be compatible. At least it would be difficult before the data sheets of the new parts are released. I suppose it wouldn't hurt to take your best guess on what the requirements would be.

Advantage of multiphase design is: lower current in each halfbridge switching stage, lower switching frequency, ripple cancelation on output LC filter. I think also TI has found this since i suppose there is already some skew between left and right channel and also between A and B half of H-Bridge. (US patent 6373336 and US patent application "20020060605 Amplifiers").

Thanks for listing these numbers!!!! :happy2:

I had done a patent and application search when I first started this project and 20020060605 hadn't been published at the time.

I highly recommend that people who are interested in Equibit designs read this! It contains a detailed description of how Equibit works (although it doesn't use that name).

It turns out that I had a couple of misconceptions about how Equibit works. Particularly regarding the way it relies on sophisticated dither to eliminate quantizing errors that would otherwise happen when using a limited 384KHz carrier to modulated 24bit data.

Fortunately, my previous misunderstandings of the internals won't affect my circuit design.:rolleyes:

Regarding power supply: I will use switching power supply with variable output voltage. Like in TACT design advantage is increased dynamic range and there is also less switching noise at low voltage and low volume.

Regarding dynamics of the power supply: switching power supplies can be made very fast. Look at the current multiphase designs for modern processors. So the best way would be to have isolated PFC conveter with large bank of capacitors on the secondary and relatively slow regulation loop (as PFC is supposed to do) and then multiphase synchronous buck converter. This would look almost exactly as already existing H-bridges and use same transistors and driver circuits.

Nice. I like the idea of circuit re-use where possible.

As I can see this project will take most of my free time next year.

Things always take longer than I expect.:blush:
Hopefully I'll have something working in a month or two.

Before I will make PCbs I would like to have some positive confirmation about timing of PWM outputs of TAS5015 chip. Is there actually no dead time between upper and lower PWM output and if there is actually time skew between left anf right channel and between left and right half of the each channel"s H-bridge?

As I mentioned above, the TAS5015 doesn't introduce any dead time compensation.
There is a small amount of time skew in the carrier of the left and right channels to reduce crosstalk from the power supply. There's very large skewing differences between the halves of the H-bridge.

What would be best SPDIF receiver chip for the modulator?

Since you're using the AD1896 ASRC, the choice of receiver chip is less critical.

I'm using a DIR1703, which allows the use of an external receiver (for better noise rejection). It's SpACT clock recovery algorithm is better than most because it isn't as data-dependant when looking at the incoming bi-phase mark stream. It is limited to 96KHz.

If you have a 192KHz source, the CS8416 would probably be a good choice. Besides the 192KHz compatibility, it has some nice features such as multiplexed inputs and automatic buffering of non-audio data. It's also probably a little bit easier to use. It's performance at 96KHz or lower isn't as good as the DIR1703, however.

Is there any advantage in using DF1704 oversampling filter?

No.
The TAS5015 has its own built-in oversampling filter.

It's nice to have you along working on this!

Regards,
Brian.:cubist:
 
Did anybody ever think of using a PCM1760 folllowed by a presettable counter to be used as a digital PWM generator (although with analog inputs) ? There would even be a few tricks possible to bring down the switching frequency and keeping the resolution the same, which could be implemented using some additional cheap standard logic ICs.

To R. McAnally:

I do not fully agree with you, regarding the stability issues for PWM amps.
If you take the feedback off before the output fillter, unconditional stability can be achieved much more easily than with ANY linear NFB amplifier type !!
If you want to include the output-filter, things start getting tricky though but still not unfeasible.
Do you know the patent # of the Hafler/Rockford amplifier ?
I wonder if their topology is really that new. I have done a lot of patent search on switching amplifiers and I must say that >95% of all the class-d patents are not covering really new ideas but are new variants of the circuit principle used within the famous Sony TA-88N amplifier (mine included :( ).
I know that TI has patents on class BD amps, where they use a multiphase bridge topology to achieve a BD - like behaviour. There are small IC amps available with this technology, intended to be used for small handheld devices where power consumption is of prime importance. These can be used without any output inductor and should still provide reasonable EMC figures (according to the manufacturer).
BTW: A real single-ended BD topology could be prone to crossover distortion.

Regards

Charles
 
Petter and Brian, thanks for the reply.

Since it seems there is interest how equibit works, I will list some of other patents:

Original patent, even before Toccata technology was formed: http://l2.espacenet.com/espacenet/bnsviewer?CY=gb&LG=en&DB=EPD&PN=WO9737433&ID=WO+++9737433A1+I+

Enhancements to the switching stage (noise filtering and current sense):http://l2.espacenet.com/espacenet/bnsviewer?CY=gb&LG=en&DB=EPD&PN=WO9959241&ID=WO+++9959241A2+I+ and http://l2.espacenet.com/espacenet/bnsviewer?CY=gb&LG=en&DB=EPD&PN=WO9959242&ID=WO+++9959242A2+I+

Negative feedback from the output stage (seems it was never implemented):http://l2.espacenet.com/espacenet/bnsviewer?CY=gb&LG=en&DB=EPD&PN=WO0046919&ID=WO+++0046919A2+I+ based on principle first conceived for analog class D:http://l2.espacenet.com/espacenet/bnsviewer?CY=gb&LG=en&DB=EPD&PN=WO9945641&ID=WO+++9945641A1+I+

Regards,

Jaka
 
Re: Damping Factor

Brian Brown said:
In real life, an Equibit amplifier has no feedback at all to be screwed up by the back-EMF


Feedback is the reason traditional linear amps are so good at controlling back EMF. This is due to the extremely low output impedance (to the rails) it creates. True, too much back emf can cause instability, but this rarely EVER happens unless you are trying to drive a capacitor.

Brian Brown said:
<class D> it has a very direct low-impedance path between the power supply and speaker that is more effective at shunting out back-EMF.


it doesnt matter if the mosfet is half way on, or fully on half of the time. plain and simple, without feedback, you have a higher impedance to the rails, and in turn less cone control and DF. add to the fact the reactive nature of the output inductor(s) and you have even more impedance. adding a reactive load in this case only lessens the DF.

Brian Brown said:
So in terms of the 'Spirit of the meaning' of damping factor, an Equibit amplifier is superior to most others.

I'm surprised that TI included this spec in the report without any explanation.



simply because the output is switching is no excuse for poor damping =). this is why TI does not try to explain why 15 is a good damping ratio. it simply is not. your explaination holds no scientific water, and is simply wrong in some areas.

So where can i find this definition of "spirit of meaning"? it seems you have casually tried to define something with no real (or true) scientific backing. a subjective approach indeed.
 
Hi phase,


the patent number you are searching is US 6,097,249. Nothing special IMHO.

I have also built my first class D amplifier more than 10 years ago and it was based on now obsolete TDA7260 from SGS-Thomson.
(self oscillating with integrator and hysteresis comparator, like sigma delta but without clock and D-flip-flop.) It was actually used in PA system driving 8 horn loaded drivers in parallel (1ohm impedance).

Regards, Jaka
 
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